| Monday December 31 2007 |
| Time | Replies | Subject |
|---|
| 11:15PM |
3 |
One Way Delay in Audio Over Analog |
| 9:13PM |
2 |
Problem with Polycom Soundpoint IP 320 Hardphone |
| 5:56PM |
0 |
How to use AddQueueMember with IAX2 peers? |
| 5:36PM |
3 |
Polycom Digit Map |
| 2:42PM |
0 |
Require IP Phones in Pakistan |
| 10:31AM |
1 |
PRI Crapping Out Regularly |
| 6:05AM |
1 |
app_echo.c |
| |
| Sunday December 30 2007 |
| Time | Replies | Subject |
|---|
| 3:48PM |
1 |
Zap channels for HFC-S PCI card not responding |
| 6:41AM |
2 |
asterisk callerid |
| 12:16AM |
1 |
Looking for PSTN provider with unlimited inbound/outbound plan |
| |
| Saturday December 29 2007 |
| Time | Replies | Subject |
|---|
| 10:50PM |
0 |
Cubix features at Asterisk: who is online |
| 6:32PM |
2 |
Cirpack KeepAlive packets causing SIP errors |
| 3:27PM |
1 |
Realtime & sip.conf |
| 1:58PM |
1 |
OT: Is Cisco 7960 SIP firmware same as 7940 SIP firmware? |
| 1:20PM |
0 |
Cisco IP phone 7975G + SCCP + Asterisk-1.4 |
| 9:19AM |
5 |
Directories Used by Asterisk |
| 8:14AM |
0 |
IAX2 failed to authenticate; it uses wrong name |
| 7:23AM |
5 |
Digium Asterisk Appliance voicemail & logs |
| 1:28AM |
8 |
Asterisk 1.4 Fax |
| 12:56AM |
1 |
Not Able To tar zxvf zaptel-*.tar.gz |
| 12:01AM |
0 |
Voicemail App |
| |
| Friday December 28 2007 |
| Time | Replies | Subject |
|---|
| 11:26PM |
0 |
Building prototype devices? |
| 11:16PM |
0 |
New voicemail vs. minivm |
| 10:10PM |
1 |
sip.conf & realtime |
| 9:20PM |
1 |
IVR help, please |
| 6:23PM |
0 |
call queuing not detecting caller hang up when call originates from voip provider |
| 6:12PM |
2 |
Problems with zaptel and HFC-S PCI card |
| 4:36PM |
1 |
Definity G3R and MWI |
| |
| Thursday December 27 2007 |
| Time | Replies | Subject |
|---|
| 10:38PM |
8 |
New voicemail app (supports many interfaces, including Audix) |
| 9:13PM |
0 |
VoIP 2008 : wish list and predictions |
| 4:33PM |
1 |
How does Asterisk scale to 500-1000 phones? |
| 4:10PM |
2 |
No SMDI interfaces are available |
| 3:07PM |
3 |
CDR |
| 2:13PM |
1 |
Samsung iDCS 500R2 <PRI> Asterisk 1.4.* |
| 10:01AM |
0 |
zap transfer |
| 9:37AM |
1 |
application not load |
| 9:32AM |
3 |
Grandtream Conference issue |
| 9:28AM |
3 |
Performance Issues Degradation After 6 Calls |
| 3:02AM |
1 |
SIP Channel jitter buffer issue |
| |
| Wednesday December 26 2007 |
| Time | Replies | Subject |
|---|
| 10:11PM |
0 |
Getting MOH after Attended Transfer |
| 10:07PM |
0 |
Fwd: Gotoif Time |
| 6:53PM |
1 |
smsq, Zaptel in UK |
| 6:06PM |
2 |
No cdr_csv after upgrade from 1.2.x to 1.4.x |
| 5:36PM |
2 |
Gotoiftime help |
| 5:07PM |
2 |
Unified Messaging On Thin Client / Terminal Server |
| 5:03PM |
1 |
ISDN BRI support with HFC-PCI cards? |
| 2:32PM |
2 |
Two lines for outgoing calls |
| 10:53AM |
0 |
autoservice.c |
| |
| Tuesday December 25 2007 |
| Time | Replies | Subject |
|---|
| 11:56PM |
1 |
Softphone to be installed on the Mobile |
| 6:50PM |
1 |
cdr_adaptive_odbc and custom rdms fields |
| |
| Monday December 24 2007 |
| Time | Replies | Subject |
|---|
| 10:20PM |
1 |
Marry Christmas and Happy New Year!!! |
| 2:29PM |
2 |
SIP Conference phones |
| 2:00PM |
1 |
sip.conf for internetcalls.com |
| |
| Sunday December 23 2007 |
| Time | Replies | Subject |
|---|
| 10:53PM |
1 |
Active Calls |
| 6:05PM |
3 |
OpenVox A800P01 and ZT_CHANCONFIG failed |
| 12:31PM |
0 |
Need some one to make a test call |
| 10:51AM |
1 |
Asterisk 1.2.26 badly broken? |
| 1:34AM |
2 |
PXE-bootable diskless Asterix distro? |
| |
| Saturday December 22 2007 |
| Time | Replies | Subject |
|---|
| 11:55AM |
1 |
Sounds transscript / speech synthesis |
| 7:51AM |
7 |
Summary: Upgrading to Asterisk 1.4 |
| 5:40AM |
5 |
call-limit in database |
| 3:32AM |
1 |
On-the-phone |
| 1:49AM |
0 |
Dead Incoming call - Sangoma A200 |
| |
| Friday December 21 2007 |
| Time | Replies | Subject |
|---|
| 10:51PM |
2 |
ODBC Voicemail and performance.... |
| 10:24PM |
0 |
Control playback |
| 8:06PM |
0 |
SIP hangup on call proceeding message |
| 7:28PM |
2 |
best way for night ringer?? |
| 5:55PM |
3 |
Polycom 330 beep on new VM |
| 3:16PM |
3 |
7970 CTLFile.tlv? |
| 2:57PM |
1 |
Send SIP 100 Trying instead of 183 Session Progress |
| 2:55PM |
1 |
Asterisk SIP handling - why 491 Request Pending response |
| 2:38PM |
0 |
Incoming CID change |
| 12:13PM |
2 |
Snom 370 buton Recordings |
| 11:30AM |
0 |
Resposta automàtica (was: asterisk-users Digest, Vol 41, Issue 67) |
| 10:41AM |
1 |
txfax not working with spandsp |
| |
| Thursday December 20 2007 |
| Time | Replies | Subject |
|---|
| 11:13PM |
0 |
Failed Call Debugging? |
| 9:38PM |
0 |
Asterisk and Chan_h323: all calls are not going |
| 9:29PM |
2 |
Telco MWI Detection on TDM400 Interface? |
| 7:53PM |
0 |
H323 and Gatekeeper |
| 7:43PM |
1 |
MeetMeConference |
| 7:01PM |
3 |
Realtime: Should I say or should I go (now) ? |
| 6:02PM |
0 |
OT: VoIP SLA for SIP trunking - SMEs |
| 3:39PM |
1 |
put fxo channel before E1 channel? |
| 3:32PM |
2 |
Cisco 7961 new firmware stops reading configuration files |
| 2:04PM |
0 |
[VOIP-Users-Conference] Re: Digium: as of this a.m., one million Asterisk downloads this year |
| 5:11AM |
1 |
Asterisk.NET API --help required |
| 4:33AM |
7 |
ip phone suggestion for Asia? |
| 1:37AM |
1 |
Asterisk 1.4.15, Solaris and record command |
| 1:12AM |
2 |
turn off auto-seek extention - force use timeout |
| |
| Wednesday December 19 2007 |
| Time | Replies | Subject |
|---|
| 10:47PM |
1 |
noun-verb vs verb-noun aka dogs black vs black dogs |
| 10:12PM |
3 |
Realtime logic in Asterisk 1.4.16.1 |
| 9:35PM |
2 |
Bulk Reverse Phone Lookup |
| 9:00PM |
1 |
Asterisk 1.4.16.1 |
| 8:33PM |
0 |
problems chanskype on ubuntu gutsy |
| 7:20PM |
5 |
Using * in extension name |
| 4:03PM |
0 |
Asterisk Realtime SIP rtcachefriends |
| 2:53PM |
1 |
IAX for asterisk to asterisk |
| 2:39PM |
2 |
asterisk with alsa |
| 2:01PM |
1 |
Asterisk and Codecs: g729 and g723 |
| 1:54PM |
2 |
Download Asterisk |
| 12:11AM |
1 |
G.278 RTP conversation capture, please. |
| |
| Tuesday December 18 2007 |
| Time | Replies | Subject |
|---|
| 9:52PM |
2 |
Asterisk/iaxclient IAX2 source port |
| 9:27PM |
2 |
resync linksys SPA9XX config file from Asterisk |
| 9:22PM |
1 |
Leading 0 in PRI outbound |
| 8:04PM |
0 |
Cisco 7970 BLF/Presence |
| 8:03PM |
0 |
AST-2007-027 - Database matching order permits host-based authentication to be ignored |
| 7:50PM |
2 |
Asterisk 1.4.16 and 1.2.26 released |
| 7:35PM |
1 |
SIP Anonymous auth |
| 7:28PM |
2 |
How to change sendmail return path |
| 7:25PM |
0 |
Softphone |
| 7:21PM |
0 |
Doorbell Siedle DCA 612 and Asterisk? |
| 6:27PM |
1 |
Call Recording on Hanup |
| 4:43PM |
1 |
Dropped Calls |
| 3:16PM |
4 |
AsteriskNOW release date??? |
| 2:20PM |
1 |
How to automaticaly close calls when Asterisk didn't receive the bye request ? |
| 12:37PM |
1 |
Asterisk GUI - Call Waiting |
| 10:47AM |
4 |
All trunk are busy please try your call again later |
| 9:59AM |
4 |
Using MysqlPool Application 1.4 |
| 2:37AM |
2 |
BLF trouble |
| 1:50AM |
0 |
Echo - when pressing digits |
| |
| Monday December 17 2007 |
| Time | Replies | Subject |
|---|
| 10:50PM |
0 |
Automatic tests (was Re: Upgrade to Asterisk 1.4 - it's one year's old!) |
| 10:28PM |
2 |
Music On Hold |
| 6:35PM |
8 |
Queue calls drop to voicemail intermittantly |
| 5:49PM |
2 |
SIP call interrupted after 64 seconds |
| 4:23PM |
0 |
Friday @12 PM EST VOIP Users Conference + Aus/NZ/India/Japan conference event |
| 1:07PM |
1 |
Mail Test |
| 11:57AM |
0 |
Problem in Recording file on Hangup ????? |
| 11:43AM |
0 |
Cannot allocate memory |
| 4:45AM |
3 |
VoIP service providers/PSTN termination points |
| 2:21AM |
1 |
dial, answered and then hangup |
| 1:53AM |
3 |
Trixbox Phones Home |
| |
| Sunday December 16 2007 |
| Time | Replies | Subject |
|---|
| 8:09AM |
1 |
Calling Party Category Field |
| 7:41AM |
0 |
LDAPget question, usage |
| 4:42AM |
0 |
Trixbox Arbitrary Command Execution Vulnerability |
| 12:55AM |
1 |
Newbie question: how to proxy the *real* caller-id on find-me/follow-me |
| 12:25AM |
1 |
Reputable company for SIP/IAX2 trunking |
| |
| Saturday December 15 2007 |
| Time | Replies | Subject |
|---|
| 5:57PM |
2 |
DNS broken for www.voip-info.org ?? |
| 12:09PM |
0 |
Open ITU G.107 Implementation to measure voice quality |
| 12:01PM |
0 |
OpenVox B800P and asterisk 1.4/ mISDN-1_1_7 |
| 10:57AM |
17 |
Upgrade to Asterisk 1.4 - it's one year's old! |
| |
| Friday December 14 2007 |
| Time | Replies | Subject |
|---|
| 8:24PM |
1 |
Asterisk to make multiple extensions simultaneous calls on a single telephone line |
| 7:43PM |
2 |
SIP fails to register |
| 6:32PM |
3 |
GUI for Asterisk: Call Flow |
| 5:26PM |
2 |
chan_h323 compilation |
| 3:26PM |
0 |
Monitor a queue |
| 1:31PM |
1 |
Asterisk Qeueu with static agent |
| 12:49PM |
2 |
Stange pause between extensions commands. |
| 11:55AM |
0 |
asterisk-users Digest, Vol 41, Issue 46 |
| 10:20AM |
1 |
ZRTP + asterisk and Best Security Practice |
| 8:06AM |
0 |
G729 on PS3 Cell |
| 6:41AM |
2 |
Poor gsm playback |
| 1:55AM |
6 |
[Zaptel] Why no port to Windos? |
| 12:30AM |
0 |
Problem with TE205P with TeleWest in the UK |
| |
| Thursday December 13 2007 |
| Time | Replies | Subject |
|---|
| 11:16PM |
0 |
Zaptel 1.2.22.1 and 1.4.7.1 released |
| 11:12PM |
0 |
Libpri 1.2.7 and 1.4.3 released |
| 9:38PM |
2 |
asterisk-users Digest, Vol 41, Issue 44 |
| 8:25PM |
1 |
Asterisk 1.2.18 and Polycom phones not forwarding anymore |
| 4:32PM |
1 |
Cell Phone SMS |
| 4:06PM |
0 |
CallManager sip trunk - callerid name? |
| 4:03PM |
1 |
DID in Cape Town South Africa |
| 12:49PM |
2 |
How do I do this? |
| 7:05AM |
1 |
chan_mobile problems |
| 6:52AM |
0 |
Fwd: Re: Re: calls are getting dis |
| 6:03AM |
1 |
calls are getting disconnected automatically |
| 1:52AM |
0 |
Didnt get a frame from Channel and call gets |
| 1:24AM |
1 |
Sipura provisioning |
| |
| Wednesday December 12 2007 |
| Time | Replies | Subject |
|---|
| 11:42PM |
1 |
Farward calls between 2 sip servers |
| 10:10PM |
1 |
Sip Version |
| 8:28PM |
4 |
TDM400 hangup issue in China |
| 6:38PM |
0 |
Can Local channels inhibit an Answer() until it is satisfied with the endpoint? |
| 6:22PM |
2 |
Linksys SPA962 with SPA932 unexpected reboots |
| 5:32PM |
1 |
Account codes in CDR |
| 4:45PM |
3 |
Polycom Paging |
| 4:14PM |
0 |
[Fwd: Request for testing SIP TCP/TLS] |
| 2:14PM |
1 |
Caller ID Issue |
| 1:01PM |
4 |
Enable/Disable Sip without registration |
| 12:30PM |
1 |
Asterisk B2BUA and Site to Site transfers |
| 12:09PM |
4 |
Call Center Setup on asterisk |
| 8:32AM |
0 |
asterisk-users Digest, Vol 41, Issue 38 |
| 8:08AM |
3 |
Load Balancing over 2 E1 Lines |
| 2:00AM |
5 |
Call Quality Issues With 2 Trixbox's - Router Issue? |
| |
| Tuesday December 11 2007 |
| Time | Replies | Subject |
|---|
| 10:14PM |
6 |
Most Stable version of Asterisk |
| 10:07PM |
1 |
Bribane bases contractor.... |
| 9:30PM |
2 |
Aastra 480i CT |
| 8:00PM |
2 |
Check if SIP user is available or not ? |
| 7:06PM |
3 |
Any phone capable of displaying real time queue statistics? |
| 6:47PM |
0 |
VPN Client with the IP Phone and what its VPNServer |
| 6:44PM |
1 |
Asterisk not sending 200 OK |
| 6:02PM |
0 |
VPN Client with the IP Phone and what its VPN Server |
| 5:51PM |
1 |
Asterisk on IBM Netvista 2800 8364-EXX? |
| 5:23PM |
2 |
OT - Fax and anti-spam |
| 5:06PM |
0 |
asterisk-users Digest, Vol 41, Issue 35 |
| 4:37PM |
0 |
new Asterisk installation with openvox 1.2 or 1.4? |
| 3:26PM |
1 |
merge gsm files |
| 3:22PM |
1 |
RFC3389 message |
| 2:30PM |
2 |
Recorded calls skipping |
| 1:03PM |
2 |
Unicall protocol error. Cause 32776 |
| 12:22PM |
2 |
Iax and ZAP |
| 12:03PM |
1 |
rollback procedure requirements before asterisk upgrade |
| 11:49AM |
5 |
hi |
| 10:09AM |
4 |
X100P Fxo card headaches |
| 8:16AM |
2 |
VPN Client with the IP Phone, and what its VPN Server |
| 8:14AM |
1 |
Asterisk and NAT |
| 8:07AM |
1 |
Video Conference Or Server |
| 7:39AM |
0 |
Monitoring Asterisk 1.4.15 |
| 3:59AM |
1 |
Appending two voice files |
| 3:30AM |
1 |
Fw: asterisk performance |
| 3:08AM |
0 |
line cut |
| 1:58AM |
5 |
SMS gateway recommendation |
| |
| Monday December 10 2007 |
| Time | Replies | Subject |
|---|
| 11:26PM |
1 |
Didnt get a frame from Channel and call gets disconnected |
| 10:04PM |
1 |
Pickup re-invite |
| 8:04PM |
0 |
Checking Dial Status |
| 7:08PM |
0 |
Cisco 7911g Poor Audio Quality w/ Asterisk Voicemail and MOH |
| 6:17PM |
2 |
asterisk 1.4 with around 230 SIP connections |
| 6:00PM |
2 |
foneBRIDGE2 vs. foneBRIDGE2-EC |
| 5:39PM |
2 |
Dynamically change sip.conf properties. |
| 5:30PM |
0 |
text to speech |
| 5:24PM |
0 |
diferents events between ast1.2 & ast1.4 ?? |
| 3:06PM |
2 |
SIP 7960 soft key customization? |
| 1:16PM |
2 |
asterisk linkedin group |
| 1:01PM |
0 |
Gateway doesn't ring |
| 12:29PM |
3 |
Graceful Asterisk Shutdown |
| 12:24PM |
2 |
Using Asterisk to connect 2 locations with legacy PBX |
| 11:46AM |
0 |
CAPI didn't get a frame | avoiding initial deadlock | multiple instances of Asterisk |
| 11:15AM |
0 |
Catalyst 2950 series with Asterisk + FAX |
| 11:10AM |
0 |
Asterisk + Cisco Call Manager Express |
| 4:59AM |
1 |
T.38 fax solution, opinions? |
| 12:55AM |
3 |
One server, multiple companies |
| |
| Sunday December 9 2007 |
| Time | Replies | Subject |
|---|
| 10:14PM |
0 |
Dual-home Wifi/GSM phones for North America |
| 9:18PM |
1 |
[DB] Using SQLite instead of AST? |
| 8:33PM |
2 |
Don't enter a queue if no one is logged in |
| 3:53PM |
1 |
Installing/configuring TE120P debian way |
| |
| Saturday December 8 2007 |
| Time | Replies | Subject |
|---|
| 3:50PM |
0 |
Asterisk CDR Variable |
| 11:50AM |
1 |
Direct Inward Dialing How To |
| 10:34AM |
0 |
Can't listen to voicemail message |
| 9:30AM |
5 |
using modem with asterisk |
| |
| Friday December 7 2007 |
| Time | Replies | Subject |
|---|
| 9:51PM |
3 |
Using XML for configuration management, single-source-of-truth, etc. |
| 8:53PM |
1 |
Problem with the ring timeout in dial command for local extensions |
| 8:04PM |
1 |
Function vs. Application? |
| 6:04PM |
1 |
Limit participants in Meetme... |
| 5:02PM |
1 |
Show calls in progress |
| 4:58PM |
2 |
Polycom 601 stops ringing |
| 4:50PM |
0 |
AMQP Support for Asterisk? |
| 4:28PM |
2 |
Open Asterisk Exchange Project |
| 4:21PM |
2 |
Sidetone with Snom 370 |
| 2:40PM |
0 |
Perspective on Asterisk |
| 10:37AM |
0 |
dtmf detection not working on sip trunks using asterisk-1.4.15 |
| 9:55AM |
4 |
Any idea how making Asterisk "transparent"? |
| 9:49AM |
1 |
Pickup cmd |
| 7:08AM |
2 |
PHP AGI script |
| 6:24AM |
0 |
Asterisk is not adding Via field |
| 5:14AM |
2 |
7960 Won't Register Yet Multiple Attempts? |
| 3:57AM |
2 |
Where does the call go in the dialplan after a call disconnects |
| 12:36AM |
0 |
PRI: calling an Unallocated Number |
| 12:09AM |
2 |
scrubbing a call list to remove cell phone numbers |
| |
| Thursday December 6 2007 |
| Time | Replies | Subject |
|---|
| 11:40PM |
2 |
Print CALLERID in CLI during "pri debug " |
| 10:26PM |
0 |
Perl FastAGI service port. |
| 10:05PM |
1 |
"Happy Birthday Asterisk" |
| 9:13PM |
1 |
Voicemail Question |
| 8:35PM |
0 |
VOIP Users Conference for Friday Dec 7 @ 12 Noon EST |
| 8:04PM |
3 |
CDR Function in Hangup Channel |
| 7:22PM |
4 |
Probems receiving 200ok message |
| 6:20PM |
3 |
Setting Multiple Values via func_odbc ...? |
| 6:18PM |
1 |
Dial() Macro option error in 1.4.15 |
| 3:37PM |
3 |
Setting custom field in CDR |
| 3:18PM |
2 |
Cisco power injector with GXP2000 phones |
| 3:00PM |
2 |
Logging in and off sessions in the dialplan |
| 2:17PM |
0 |
Increasing the voice volume for digium card |
| 2:11PM |
2 |
Call Center Scenario -- take 2 |
| 1:19PM |
3 |
Play Beep instead of MOH |
| 1:07PM |
0 |
IVR problem (Trixbox) |
| 11:30AM |
1 |
DeadAgi |
| 10:26AM |
0 |
Polycom call drops |
| 9:53AM |
2 |
GSM and CDMA Gateways |
| 9:47AM |
3 |
asterisk performance |
| 9:44AM |
0 |
C&W ISDN110 |
| 6:33AM |
0 |
Can Asterix seperate the signalling and the media ip's with Quintum |
| 3:47AM |
1 |
Running AGI script if condition met? |
| 2:51AM |
0 |
TDm804B |
| 2:33AM |
1 |
s, CDR and NoCDR in v1.4.10.1 |
| 1:19AM |
1 |
Polycom Soundpoint (NO LINE) |
| 12:49AM |
2 |
astunicall-1.2.21.0.1 packages and Sangoma A104D - ERROR |
| 12:34AM |
0 |
astunicall-1.2.21.0.1 packages and Sangoma A104D |
| |
| Wednesday December 5 2007 |
| Time | Replies | Subject |
|---|
| 10:42PM |
1 |
Cisco 7960 to 2 SIP servers? |
| 10:05PM |
8 |
Strange ISDN-problem with incoming calls out of the same city |
| 8:48PM |
4 |
Asterisk server and DSCP QOS |
| 7:25PM |
3 |
No timezone in Voicemail email? |
| 7:16PM |
0 |
Asterisk and TDM400P |
| 4:45PM |
7 |
Asterisk SIP Microsoft Outlook Integration |
| 3:47PM |
3 |
Adtran supervision problems |
| 3:45PM |
0 |
redirected call failure |
| 3:35PM |
0 |
New feature: calling all bug marshals |
| 3:18PM |
1 |
SIP-Realtime and sip reload |
| 2:57PM |
0 |
Bad behaviour between X-Lite 3.0 and Asterisk |
| 12:38PM |
2 |
New feature: calling all bug marshals |
| 9:25AM |
2 |
Increasing the voice volume from the digium card |
| 9:13AM |
2 |
Text-To-Speech synthesizer--help required |
| 8:49AM |
1 |
[HELP] Problems with VOIP organization |
| 8:40AM |
1 |
Disturbance "noise" in the background for digium card |
| 7:23AM |
2 |
Multiple contacts. |
| 6:03AM |
5 |
New feature: calling all bug marshals |
| 5:09AM |
1 |
[Fwd: load test zap channels (in and out)] |
| 5:03AM |
0 |
Melbourne Asterisk meetup - again |
| |
| Tuesday December 4 2007 |
| Time | Replies | Subject |
|---|
| 9:55PM |
0 |
Which rackable server for TDM2400 ? |
| 8:17PM |
1 |
Fax on asterisk |
| 7:34PM |
1 |
Call center scenario |
| 6:17PM |
4 |
Echo cancellation and DTMF from the Asterisk console? |
| 4:24PM |
1 |
New User |
| 4:04PM |
1 |
Soundcard necessary on an asterisk server toget output of playback()?? |
| 3:14PM |
1 |
Voicemail box |
| 2:41PM |
0 |
System Specifications for 60 Extensions |
| 2:39PM |
0 |
Digium Support Comparison with Sangoma |
| 2:02PM |
2 |
pstn call waiting and zap |
| 11:39AM |
4 |
enable eyeBeam to accept only one call |
| 9:43AM |
1 |
Gtalk callerID |
| 9:29AM |
1 |
Explain AGI and AMI |
| 7:21AM |
1 |
Problem forwarding voicemail messages |
| 4:49AM |
0 |
Queue App - (1.4.15) free agents with callers waiting |
| 4:35AM |
0 |
Queue App - crash (1.4.15) |
| 4:13AM |
3 |
Phone with public address functionality |
| 2:18AM |
1 |
IBM x3400 w/ Digium TE220 |
| |
| Monday December 3 2007 |
| Time | Replies | Subject |
|---|
| 10:10PM |
0 |
Asterisk and Ekiga Chat |
| 8:16PM |
0 |
Adhearsion Install Fails. |
| 6:20PM |
2 |
Hoteling |
| 6:14PM |
1 |
MWI error |
| 6:01PM |
4 |
Soundcard necessary on an asterisk server to get output of playback()?? |
| 5:25PM |
3 |
Anyone here using JUNGHANNS.net douBRI 2.0 ISDN ? |
| 5:04PM |
0 |
Click2call from Thunderbird |
| 3:14PM |
2 |
Red Alarm TE420 with E1s - R2 |
| 2:38PM |
3 |
Replacing Skype with Asterisk Peering Servers - and Security |
| 10:16AM |
2 |
Problem: Using timelimit (L) and Macro (M) in Dial from AGI |
| 9:34AM |
1 |
SPA-3102 Registration Failed .. need advise |
| 9:19AM |
3 |
Underground Asterisk Command Set? |
| 9:16AM |
0 |
get REMOTE SIP extension status without calling it |
| 8:11AM |
2 |
MeetMe Conference on Asterisk-1.4.13 |
| 7:03AM |
1 |
New VICIDIAL astGUIclient Release: 2.0.4 |
| 6:48AM |
1 |
Oracle and asterisk |
| 3:41AM |
0 |
Asterisk 1.4.15 sip.conf register |
| 2:02AM |
1 |
Subject: Newb Question |
| |
| Sunday December 2 2007 |
| Time | Replies | Subject |
|---|
| 10:52PM |
1 |
setting up two asterisk server as ss7 back to back. |
| 9:42PM |
2 |
Softswitch digim |
| 8:18PM |
2 |
Asterisk install beta testing/config help |
| 7:48PM |
0 |
When calling in via AGI, gsm sound file plays but sometimes drops out |
| 7:22PM |
1 |
T1 Timing Troubleshooting |
| 6:22PM |
1 |
What is the status and future of chan_mobile |
| 5:38PM |
0 |
Dictaphone Freedom interface to Asterisk ABE |
| 4:49PM |
2 |
Answering Machine Detection |
| 2:30PM |
1 |
Asterisk on Solaris |
| 1:51PM |
4 |
get SIP extension status without calling it |
| 11:05AM |
1 |
DID provider |
| 5:35AM |
1 |
Answer Machine/Fax/modem detection |
| 12:09AM |
2 |
Requiring a login to a phone |
| |
| Saturday December 1 2007 |
| Time | Replies | Subject |
|---|
| 5:50PM |
2 |
Increasing the voice volume from the diguim cards |
| 3:43PM |
2 |
cdr_pgsql error in 1.4.15 |
| 1:35PM |
0 |
Asterisk 1.4.15 Voicemail |
| 1:22PM |
1 |
Consulting/Integration Services Non-US & US *u |
| 5:42AM |
1 |
Asterisk & Cisco calling Name |
| 5:13AM |
1 |
REFER mesage extraction using SIP_HEADER |
| 2:40AM |
1 |
Registration state: Failed |